High efficiency coding of audio signal allows for compressing the sound quality that corresponds to a CD (Compact Disk) into a data amount of approximately 1/10 to 1/20 that of the original CD, by using the mechanism of human hearing. Currently, products using such technologies are distributed in the marketplace, thereby allowing for recording on a smaller recording medium and delivering via a network, for example. Main hearing characteristics used in such high efficiency coding of audio signal is simultaneous and temporal masking.
Simultaneous masking is a hearing characteristic that, in a case where sounds at different frequencies exist at the same time, when there is a small-amplitude sound in the neighborhood of the frequency of a large-amplitude sound, the small-amplitude sound is masked and becomes hard to perceive.
On the other hand, temporal masking is a masking effect in the temporal direction, and is a hearing characteristic that, for example, a small-amplitude sound existing at a time before or after a large-amplitude sound is masked to be hard to perceive. There are two phenomena for the temporal masking: forward masking where temporally-before generated sounds mask temporally-after generated sounds; and backward masking where a temporally-after generated sounds mask temporally-before generated sounds. It is known that forward masking is effective for a period in the order of several tens of msec (milliseconds) while backward masking is effective for an extremely short period of approximately 1 msec.
In a typical high efficiency coding method for audio, after orthogonally transforming a time signal by MDCT (Modified Discrete Cosine Transform), normalization is performed on the obtained MDCT coefficients on a frequency axis for each set of a plurality of MDCT coefficients (hereinafter referred to as “quantization unit”), and then, quantization and coding are performed.
The range of a quantization unit is generally narrow in the lower region and wide on the higher region, which allows for changing the number of quantization steps adaptively for each quantization unit and controlling the generation of quantization noise acceptable to the hearing properties.
However, in the above-described method, the band in which frequency components are quantized is fixed. Thus, for example, when spectra concentrate in a specific narrow band, in order to quantize the spectral components thereof with sufficient accuracy, many bits should be allocated to many spectra included in the same quantization unit as one of these spectral components.
Generally, compared to quantization noise added to a signal where energy is evenly distributed over a wide frequency band, quantization noise in a tonal signal in which energy concentrates on a specific frequency thereof may be extremely loud to the ear, and result in a great acoustical disturbance. Further, without a sufficient accuracy for quantizing tonal components, when the spectrum component thereof are converted back into a signal on the time axis and combined to the preceding or following blocks, the distortion between the blocks will be large, resulting also in a great acoustical disturbance.
Thus, in order to code a tonal component, quantization should be performed with sufficient number of bits. It is needed to perform quantization by allotting many bits to many spectra in a quantization unit which include a tonal component when quantization accuracy is determined for each predetermined band as described above, thereby the coding efficiency decreases.
For an approach to solving such a problem, in Japan Patent No. 3336617 there is provided efficient coding even for a tonal signal by separating a frequency component into a plurality of signal components and coding them separately.
Meanwhile, in such high efficiency coding of audio signal, significant amounts of computation and memory are needed for coding/decoding. Therefore, in a case where once a simple signal processing is performed on a coded code string, a desired signal processing can be performed with a small amount of computation and a small memory, by directly changing the parameter of the code string, instead of by decoding and subsequently performing a desired signal processing and then re-coding.
Japan Patent No. 3879249 discloses an invention for enabling filtering of a signal by directly changing normalization coefficient information in a code string. Also, Japan Patent No. 3879250 discloses an invention for enabling level adjustment of a signal by directly changing normalization coefficient information in a code string.
[Patent Document 1] Japan Patent No. 3336617
[Patent Document 2] Japan Patent No. 3879249
[Patent Document 3] Japan Patent No. 3879250